similar to: 1.6.2.8: need sip reload to reach peers.

I've just got Uverse installed. I had dsl, but ATT insisted I couldn't keep my old dsl, but had to switch to Uverse internet - vdsl. My setup: linux box as router : 10.10.11.252 asterisk box: 10.10.11.180 10.10.11.252 is multihomed and connected to the Uverse Residential Gateway. I've set it up as DMZplus, and it shows the public ip address as eth1. I can ssh into the

My internal lan is small, 100mb, all wired. aastra phones. sip show peers ....... 142/... 10.10.10.42 D A 5060 OK (136 ms) 144/... 10.10.10.44 D A 5060 OK (138 ms) 145/... 10.10.10.45 D A 5060 OK (133 ms) But pings are < 1ms: ping 10.10.10.42 ........ rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms Why are the sip latencies so

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP

I have a 100MB internal lan. aastra's are wired. asterisk box is wired next to the switch. But look: sip show peers ........ 142/142 10.10.10.42 D A 5060 OK (137 ms) 144/144 10.10.10.44 D A 5060 OK (136 ms) 145/145 10.10.10.45 D A 5060 OK (168 ms) 150/150 10.10.10.50

We are looking for SIP trunks for our * pbx for our business. Being able to port our numbers is an absolute requirement. teliax can do it, but I am unsure of the others. Anyone have experiences (good, bad) with the above mentioned providers to share? Eg reliability, quality, etc. -Dan

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to allow access to the office from home. But the ip provider (cablevision) blocks udp 5060. I can see the register packets leaving on wireshark, but nothing received by office. Changed to port to 6111 and now the packets show up. In the server I've set port=6111 in the device in sip.conf, but * is NOT listening for 6111: netstat -an | grep 5060 tcp 0 0

I've tried to create a subroutine that sets callerid name based on number. extensions.conf: ........... exten => s,1,Answer() exten => s,n,GoSub(set-callerid-name,0${CALLERID(num)},1) exten => s,n,Dial(${mainline},60) ....... [set-callerid-name] exten => 0,1,NoOp( no CALLERID num set) exten => 02025462677,1,Set(CALLERID(name) = "Fred" ) ................ exten =>

I'm having trouble setting callerid with teliax. I use a simple dial-out subroutine to set the callerid depending on the calling extension, and then dial out. Teliax is saying they're not seeing any callerid info. [DialOut] ; subroutine for dialing out. exten => s,1,NoOp(Context: DialOut called with outgoing number ${ARG1} ) exten => s,n,NoOp(${CALLERID(num)}XXXX) exten =>

Hello everybody, I am having a problem with realtime SIP peers. On Asterisk 1.8, I had SIP peers for external SIP providers configured in database and additional register lines in sip.conf so they would register. Now I upgraded to Asterisk 11.3.0, partly because of the promised callbackextension feature for realtime peers (https://reviewboard.asterisk.org/r/1717/). Removed the 'register'

Hi, list. We have a problem with loss peers after 'sip reload', our configuration: Asterisk 11.6-cert1, SIP realtime peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do 'sip reload' , peers are removing from available. Before `sip reload` : srv-pbx2*CLI> sip show peers Name/username Host

I am trouble finding a configuration that works for ChanIsAvail and SIP. My two providers are Voxee and Teliax. I have these lines in a macro exten => s,n,ChanIsAvail(SIP/teliax&SIP/voxee) exten => s,n,Cut(CH=AVAILCHAN,-,1) exten => s,n,NoOp(AVAILCHAN= ${CH}) ; Dial the available Channel exten => s,n,Dial(${CH}/${ARG1},60,t) Looking at the execution, I can see what the AVAILCHAN

I've have a simple caller id lookup on incoming: [teliax-in] .......... exten =>s,n,GoSub(set-callerid-name,0${CALLERID(num)},1) ................ [set-callerid-name] exten => 0,1,NoOp( no CALLERID num set) exten => 02135590993,1,Set(CALLERID(name)=Matthew ) ............................................... exten => _0!,n,NoOp(CALLERID: ${CALLERID(name)}) exten => _0!,n,Return()

I know that the wiki has an extensive list of European VoIP providers out there....but there's so many that it's kind of hard to sort through. So I was wondering if anyone could recommend some reliable SIP/IAX termination providers in Europe? Something like VoicePulse Connect, NuFone, Vitelity, or Junction Networks based out of Europe. I really don't trust a US VoIP company for

Apparently we've been having calls sporadically drop. We're using an IAX outbound trunk and SIP adapters on the inside. Below is a log excerpt detailing one of the calls which dropped, and it looks largely normal to me except for this: Jan 5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel: IAX2/teliax-2 Jan 5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging

We are setting up asterisk with Teliax and having trouble getting the incoming call to work all the time, the outgoing does not seem to have a problem. I have worked with their support but since they say that we are getting the initial call to our server they want to charge to take a look. They did a tcpdump and we are seeing an attempt but no CLI most of the time. Some times we see this but it

Many of my dialplan scenarios involve transferring incoming calls back out to other numbers. For reasons of call quality and bandwidth, I would like for the calls to be reinvite'd to bypass my server with the audio channel. What I am seeing is that my server does indeed send the reinvites, and I get OK responses, but the audio never stops passing through my server. I've been fooling